[..] |
ASTERISK-04000: [patch] fix peer matching for multiple peers at the same IP address in 'insecure' mode |
ASTERISK-04001: [patch] Add strftime() function |
ASTERISK-04002: [patch] Wrong interpretation h323.conf |
ASTERISK-04003: new option in app_dial ('n') collides with privacy option... |
ASTERISK-04004: [patch] transcode via sln breaks chan_local |
ASTERISK-04005: [patch] cli auto-completion after unloading chan_iax causes core |
ASTERISK-04006: [patch] Incorrect indenting, possible bug |
ASTERISK-04007: DTMF features do not work for calling party |
ASTERISK-04008: progressinband=yes does not work |
ASTERISK-04009: The AgentLogin Applicaction don't report the agent number in the CDR Record |
ASTERISK-04010: [patch] Raise write() failure in my_zt_write from Debug to Warning |
ASTERISK-04011: [patch] improved chan_oss.c driver |
ASTERISK-04012: [patch] Format change of app_sms |
ASTERISK-04013: [patch] Changes of %i for %d |
ASTERISK-04014: [patch] Forwarding Context |
ASTERISK-04015: Call signalling failure connecting to Cisco Callmanager 4.0.1 using h.323 |
ASTERISK-04016: [patch] Create manual page in various formats |
ASTERISK-04017: [patch] Create manual page in various formats |
ASTERISK-04018: [patch] Raise write() failure in my_zt_write from Debug to Warning |
ASTERISK-04019: [patch] changes to build system to remove run-time use of hardcoded paths (and other improvements) |
ASTERISK-04020: [patch] pri_timer2idx usage incorrect in chan_zap.c |
ASTERISK-04021: Voicemail prepend did not work |
ASTERISK-04022: [patch] Change debugging level |
ASTERISK-04023: [patch] pri_timer2idx usage incorrect in chan_zap.c |
ASTERISK-04024: Asterisk stop respond and full log with "Failed to grab lock, trying again..." |
ASTERISK-04025: [patch] convert app_md5 to functions and fix an ast_separate_app_args bug |
ASTERISK-04026: [patch] ENUM - properly handling of multiple NAPTR records |
ASTERISK-04027: [patch] FreeBSD compile problem due to wrong header files |
ASTERISK-04028: [patch] Incorrect parsing when loading peer from ast_db |
ASTERISK-04029: [patch] pbx_wilcalu unhandled condition on poll() |
ASTERISK-04030: [patch] rtp.c: wrong error message on recvfrom error |
ASTERISK-04031: [patch] unused INSTALL_PREFIX passed to the compiler Makefile |
ASTERISK-04032: [patch] unused INSTALL_PREFIX passed to the compiler Makefile |
ASTERISK-04033: [patch] main Makefile FreeBSD-specific paths |
ASTERISK-04034: Inbound DTMF to Meetme does not work |
ASTERISK-04035: [patch] add groupcount functions |
ASTERISK-04036: [patch] documentation on file descrptors |
ASTERISK-04037: One way audio between CCM and Asterisk |
ASTERISK-04038: Wrong comment for ast_sched_add |
ASTERISK-04039: [request] possible suggestion for udev and zaptel |
ASTERISK-04040: [patch] regseconds is never non-zero in realtime |
ASTERISK-04041: Wrong comment for ast_sched_add |
ASTERISK-04042: query loop and asterisk crash with realtime iax |
ASTERISK-04043: outbound calls fail after 05/01 cvs |
ASTERISK-04044: DNID is not always set on incoming BRI calls with HFC driver bristuff-0.2.0-RC8 |
ASTERISK-04045: [patch] SetVar paradox |
ASTERISK-04046: [patch] Doxygen updates. |
ASTERISK-04047: [patch] astobj.h documentation |
ASTERISK-04048: ast_copy_string cuts off the last character when passing in a size |
ASTERISK-04049: chan_sip rejects calls with caller id containing , |
ASTERISK-04050: [patch] Fixups for chan_mgcp |
ASTERISK-04051: [patch] SIP does not close socket on unload |
ASTERISK-04052: [patch] SetTransferCapability returns a multi-line string from the description() module routine |
ASTERISK-04053: [patch] Authentication support for SIP NOTIFY requests |
ASTERISK-04054: [patch] SIP does not close socket on unload |
ASTERISK-04055: [patch] Doxygen updates. |
ASTERISK-04056: [patch] Fixups for chan_mgcp |
ASTERISK-04057: Zaptel/TE405P broken |
ASTERISK-04058: [patch] rtp.c: wrong error message on recvfrom error |
ASTERISK-04059: Add "silent" option to VMAuthenticate |
ASTERISK-04060: [patch] frame.c fix ast_frisolate for non-malloced data, comment out broken and unused functions |
ASTERISK-04061: playback(msg|noanswer) does not work |
ASTERISK-04062: [patch] tailqueues to queue frames (O(1) instead of O(n) cost) |
ASTERISK-04063: oSIP clients cannot call through Asterisk that uses non-default SIP port |
ASTERISK-04064: SEGV with Realtime and IAX2 |
ASTERISK-04065: [patch] new jb - prevent late voice frame from ending silent state |
ASTERISK-04066: [patch] g.729 codec one way audio in chan_h323 with Cisco CCM |
ASTERISK-04067: [patch] Dial Zap trunk example |
ASTERISK-04068: [patch] Dial Zap trunk example |
ASTERISK-04069: [patch] new jb - use defines for static adjustments |
ASTERISK-04070: [patch] macros for simpler string handling |
ASTERISK-04071: [patch] Several more fixes for MGCP, including a codec parsing error |
ASTERISK-04072: [patch] Volume Control |
ASTERISK-04073: [patch] Commments for channel.c /channel.h |
ASTERISK-04074: [patch] Formatting of chan_sip.c |
ASTERISK-04075: Voicemail volume on calls originating from pstn is very low |
ASTERISK-04076: [support script] this script alows a person to grab a core of the running or deadlocked process |
ASTERISK-04077: [patch] chan_iax2 - add a netstats manager command |
ASTERISK-04078: [patch] Zaptel Init Updates |
ASTERISK-04079: cdr_mysql problems with FreeBSD |
ASTERISK-04080: latest cvs fails to compile on Linux x86_64 (RHEL4) |
ASTERISK-04081: BUild Busted due to changes that cause make to fail on make version 3.79. 3.80 works |
ASTERISK-04082: Mantis isn't behaving right! |
ASTERISK-04083: [patch] add newline to res_monitor warning |
ASTERISK-04084: [patch] wrong path in config.c |
ASTERISK-04085: [patch] More proper locking, formatting fixes, etc. - C part only |
ASTERISK-04086: zttool reports incorrect clocking |
ASTERISK-04087: Sound using Monitor(), No Sound Using One Touch Record |
ASTERISK-04088: [request] Call Transfer no getting variables |
ASTERISK-04089: Cannot view any bugs unless logged into mantis |
ASTERISK-04090: [patch] add TOUCH_FORMAT channel var for OTR |
ASTERISK-04091: [patch] make language used in 'iax2 trunk debug' more correctly represent data shown |
ASTERISK-04092: [patch] add more knobs to iax2 peer qualification system |
ASTERISK-04093: [patch] tweak zapata.conf to explain purpose of 'jitterbuffers' directive and its relationship to EAGAIN errors. |
ASTERISK-04094: [patch] Some cleanups for C++ part |
ASTERISK-04095: Crash on SIP Register |
ASTERISK-04096: [patch] Two minor fixes for functions |
ASTERISK-04097: [patch] Fixes in say.c for french. |
ASTERISK-04098: [patch] gethostname may return unterminated strings |
ASTERISK-04099: [patch] App_transfer: Can't transfer without tech |
ASTERISK-04100: [patch] Additional group functions |
ASTERISK-04101: [patch] convert app_db to dialplan functions |
ASTERISK-04102: stale code in pbx.c ? |
ASTERISK-04103: Small fix to make app_directory work on solaris |
ASTERISK-04104: [patch] fix to make app_chanspy work on solaris |
ASTERISK-04105: multiple ast_mutex_unlock(&hintlock) in pbx.c::ast_extension_state_add() |
ASTERISK-04106: Small fix to make app_directory work on solaris |
ASTERISK-04107: [patch] unreachable code in chan_features.c |
ASTERISK-04108: [patch] chan_features.c remove unused argoment to indexof() |
ASTERISK-04109: Configuration script |
ASTERISK-04110: [patch] add pri show debug command |
ASTERISK-04111: IAX plaintext authentication is fubared when asterisk is compile with -march=pentium4 and -O3 |
ASTERISK-04112: [patch] Change domain length to MAXDOMAINLEN in chan_sip |
ASTERISK-04113: [patch] Formatting changes of chan_sip.c |
ASTERISK-04114: [patch] Implement more hangupcauses in SIP |
ASTERISK-04115: Portability fixes and some cleanup for utils dir |
ASTERISK-04116: [patch] Fix memory leak in numerous codec_*.c |
ASTERISK-04117: [patch] Fix reload of res_features.so when using astmm |
ASTERISK-04118: RealTime mysql does not parse commas |
ASTERISK-04119: [patch] Functions to replace SetLanguage, AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetCIDNum, SetCIDName, SetRDNIS |
ASTERISK-04120: [patch] DNID is empty on incoming "immediate" calls |
ASTERISK-04121: [patch] fix one-way audio in iax trunking |
ASTERISK-04122: adjustment to MAX_TIMESTAMP_SKEW to work around chan_zap/zaptel bug |
ASTERISK-04123: [patch] Fix 2 memory leaks in config.c which also uncovers potential problem in config file processing |
ASTERISK-04124: [request] add "agentcallbacklogin" CLI command |
ASTERISK-04125: [patch] Fix smsq compiling on older linux distros |
ASTERISK-04126: zaptel driver not returning audio data when sending DTMF |
ASTERISK-04127: [patch] Swedish grammatics |
ASTERISK-04128: Cross-compilation for PowerPC fails on codecs/gsm |
ASTERISK-04129: [patch] Define maximum number of messages per folder in voicemail.conf |
ASTERISK-04130: [patch] make clean fails to delete stereorize |
ASTERISK-04131: [patch] Swedish indications.conf.sample |
ASTERISK-04132: [patch] Counter for show agents |
ASTERISK-04133: New H323 channel driver for asterisk using ooh323c |
ASTERISK-04134: [patch] New H323 channel driver for asterisk using ooh323c |
ASTERISK-04135: codec_speex.c preventing VAD in CBR mode - very minor code change required |
ASTERISK-04136: [patch] [post 1.2] Remote MWI over IAX2 (proposal) |
ASTERISK-04137: [patch] Fix a bunch of the more useful gcc4 warnings. |
ASTERISK-04138: [request] qualify=yes in [general] does not work |
ASTERISK-04139: [patch] log2 shadows builtin floating point function. |
ASTERISK-04140: G.729A SIP handshake succeeds even without no free G.729A licenses |
ASTERISK-04141: Unable to dial out on UK PRI |
ASTERISK-04142: [patch] add cli jitter test commands |
ASTERISK-04143: probable bug (index out of range) in chan_features.c:features_new() |
ASTERISK-04144: * deadlocks after att. transfer into queue |
ASTERISK-04145: [patch] Voicemail Synchronization Issue |
ASTERISK-04146: Outbound PRI calls rejected (switchtype=national) |
ASTERISK-04147: libpri changes block all outgoing calls |
ASTERISK-04148: chan_zap.c doesn't compile |
ASTERISK-04149: [patch] new jb - improve scheduling and jitter handling in new jitter buffer |
ASTERISK-04150: Outbound calls to cell phones or other switches which do not immediately know the state of the called station hang indefinitely |
ASTERISK-04151: [patch] new jb - fix when queue_put puts frames at the head of the queue |
ASTERISK-04152: Increasing delay over time on non-Zap channels in MeetMe |
ASTERISK-04153: [patch] Speex VBR fix, VAD enable fix and VBR/ABR DTX enable fix |
ASTERISK-04154: [patch] macros for simple list navigation |
ASTERISK-04155: [patch] functions to register/unregister multiple cli entries |
ASTERISK-04156: [patch] make WaitForSilence return value meaningful |
ASTERISK-04157: [patch] iax: explicit source address selection in peer declaration |
ASTERISK-04158: cdr.c::ast_cdr_unregister() incorrect |
ASTERISK-04159: ast_readstring for res_perl. |
ASTERISK-04160: [patch] fix logging to syslog |
ASTERISK-04161: [patch] Correctly retrieve the contact address during registration |
ASTERISK-04162: [patch] Changing some CLI log messages, adding hold status support |
ASTERISK-04163: [patch] Combine config parsing stuff together |
ASTERISK-04164: [patch] RTP Padding |
ASTERISK-04165: [patch] better ast_channel_walk_locked() and ast_get_channel_by_name_locked() |
ASTERISK-04166: crash during user register request when using realtime |
ASTERISK-04167: likely bug in [undocumented] handling of fout field of ast_channel |
ASTERISK-04168: [request] MySQL table structure for Realtime |
ASTERISK-04169: [patch] [post 1.2] MeetMe cleanup, redesign process |
ASTERISK-04170: [patch] RTP Padding |
ASTERISK-04171: deadlocks when manager connection dies without sending disconnect |
ASTERISK-04172: [patch] Fix smsq compiling on older linux distros |
ASTERISK-04173: Typo in recent CVS commit (chan_iax2.c rev 1.281) |
ASTERISK-04174: [support script] this script alows a person to grab a core of the running or deadlocked process |
ASTERISK-04175: [request] possible suggestion for udev and zaptel |
ASTERISK-04176: DIALSTATUS is returning CONGESTION when it should return CHANUNAVAIL when dialing disconnected SIP phone |
ASTERISK-04177: [patch] convert app_eval to a dialplan function |
ASTERISK-04178: [patch] convert app_chanisavail to a dialplan function |
ASTERISK-04179: [patch] convert sample extensions.conf to use new dialplan functions |
ASTERISK-04180: codecs (allow/disallow) specified in database do not work with realtime |
ASTERISK-04181: [patch] Eliminate hidden copying when passing structures as arguments, small memory leak fix |
ASTERISK-04182: [request] A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel. |
ASTERISK-04183: [patch] RTP.conf and rtp.c documentation |
ASTERISK-04184: [patch] Voicemail Duration Incorrect |
ASTERISK-04185: [Patch] Make PRI_EVENT_KEYPAD_DIGIT work in chan_zap.c |
ASTERISK-04186: memory leak in chan_sip/mysqlsipfriends |
ASTERISK-04187: [patch] Voicemail Duration Incorrect |
ASTERISK-04188: CLI Does not get output from issued commands, freezes on TAB after 4053 |
ASTERISK-04189: [patch] Improve HOLD support in SIP |
ASTERISK-04190: "delayreject=yes" breaks "trunk=yes" |
ASTERISK-04191: Extreme jitters and sounds garbling |
ASTERISK-04192: [patch] "set verbose 1" does implicit "set debug 1" |
ASTERISK-04193: [patch] new jb - limit the number of contiguous interpolation frames returned by the jb |
ASTERISK-04194: [patch] Native support for OGG/Vorbis |
ASTERISK-04195: [patch] [post 1.2] Bridge of 2 existing channels including Manager API |
ASTERISK-04196: [patch] MOH turns off before 2nd user with intro enters conference |
ASTERISK-04197: While rinning Festival application: runs 1,3,5,.. produce valid sound and runs 2,4,6,... produce only noise |
ASTERISK-04198: ASTCC - Bug in Local Dial Feature - Patch Included |
ASTERISK-04199: [patch] ztdummy accuracy improvements on kernel 2.6 |
ASTERISK-04200: [patch] For broken for CLI in certain countries [au uk nz sg in za] |
ASTERISK-04201: [patch] chan_sip.c does not compile |
ASTERISK-04202: [patch] Voicemail Web Application Synchronization |
ASTERISK-04203: When making or recieving a call a click is heard a short time after the connection is made. |
ASTERISK-04204: [patch] Old style of app,appdata not parsed correctly when appdata contains a ( |
ASTERISK-04205: [patch] The IF func has a bug |
ASTERISK-04206: [Patch] Extra LOCAL_USER_REMOVE in VMAuthenticate |
ASTERISK-04207: [patch] allows for silence suppression |
ASTERISK-04208: [patch] Add IAXPEER Function |
ASTERISK-04209: [patch] new jb - patch to support resyncing the jb on dratic delay changes |
ASTERISK-04210: [Feature request] Can chan_iax pass back fault specific PRI cause codes when network errors occur? |
ASTERISK-04211: setting rtptimeout per peer basis does not work (probalby rtpholdtimeout, rtpkeepalive too) |
ASTERISK-04212: Dictate - incorrect filenames |
ASTERISK-04213: MeetMe - Save path for recordings [suggestion] |
ASTERISK-04214: port in sip.conf is now searched as bindport |
ASTERISK-04215: [patch] s extension is not used on incoming overlap call without extension |
ASTERISK-04216: memory leak in frame/channel |
ASTERISK-04217: [patch] new jb - prevent excess scheduling work |
ASTERISK-04218: [patch] option_nosupport to allow you to turn off various idiot warnings. |
ASTERISK-04219: [request] Add Support For Sending DTMF As LIMIT_WARNING_FILE |
ASTERISK-04220: [patch] Make IF func handle whitespace |
ASTERISK-04221: [patch][post 1.2] ALIAS |
ASTERISK-04222: [patch] "set verbose 1" does implicit "set debug 1" |
ASTERISK-04223: setting rtptimeout per peer basis does not work (probalby rtpholdtimeout, rtpkeepalive too) |
ASTERISK-04224: [patch] Add Auto Member Logout Support |
ASTERISK-04225: [patch] Add password to the VocemailMain function parameters |
ASTERISK-04226: [PATCH] crash/core dump at INVITE |
ASTERISK-04227: fork_cdr gives wrong cdr data |
ASTERISK-04228: [patch] Formatting changes |
ASTERISK-04229: deadlock when using internal queues |
ASTERISK-04230: [patch] Alphabatize the brief docs |
ASTERISK-04231: [patch] Make capabilities to be connection-specific rather than endpoint-specific |
ASTERISK-04232: [new functions] LET / SET |
ASTERISK-04233: [patch] Handle call options more globally |
ASTERISK-04234: [request] Enable debug on by default if set in logger.conf |
ASTERISK-04235: problem to register by sipgate.de |
ASTERISK-04236: CVS-HEAD patch to allow compilation on Solaris |
ASTERISK-04237: [patch] Add t Option To VMAuthenticate to skipuser transparently using supplied mailbox |
ASTERISK-04238: [patch] *CLI> sip debug No such command |
ASTERISK-04239: [patch] new jb - prevent reset from clearing jb settings and fix error/warning output |
ASTERISK-04240: [PATCH] REGISTER "deadlock" between SPA's and Asterisk/Non-SPA Interoperability |
ASTERISK-04241: Media is not flowing between two asterisks (Request uri missing in ACK ) |
ASTERISK-04242: [patch] ast_h323.cpp optimizations and tweaks |
ASTERISK-04243: zaptel distinctive ring detection occsionally fails |
ASTERISK-04244: relay PROGRESS messages with cause code IE |
ASTERISK-04245: [patch] stop playing beep when starting to spy on a channels and the 'q' option is supplied |
ASTERISK-04246: [request] 'indication save' from the Asterisk CLI |
ASTERISK-04247: [patch] adds AST_MAX_ACCOUNT_CODE def to cdr.h |
ASTERISK-04248: [patch] Support of Supported: and Required: headers |
ASTERISK-04249: SayAlpha stops on unknown character |
ASTERISK-04250: [patch] >Parses country code from a dial string (Useful for billing) |
ASTERISK-04251: [patch] various patches for a cleaner compile with -Werror |
ASTERISK-04252: [patch] add new realtime function realtime_direct |
ASTERISK-04253: zaptel makefile cleanup |
ASTERISK-04254: [patch] Fix help so it prints warning if no help text is provided |
ASTERISK-04255: Sending a test message FROM a SIP phone gives "405 Method Not Allowed" |
ASTERISK-04256: zaptel/Makefile broken for RHEL3 |
ASTERISK-04257: [patch] SIP timer support |
ASTERISK-04258: problem with dynamic ip-address |
ASTERISK-04259: mp3 music on hold use many cpu-time without need |
ASTERISK-04260: [patch] new config option 'rtautoreg' |
ASTERISK-04261: [patch] Clean up some message handling |
ASTERISK-04262: [PATCH] making H.323 noSilenceSuppression work |
ASTERISK-04263: Echo problem with TDM400P (4 fxo, ISRAEL mode, tone zone 19) |
ASTERISK-04264: Code bug in chan_iax2.c |
ASTERISK-04265: [request] mpg123 should only be spawned once per music on hold class |
ASTERISK-04266: Fax detection will not properly jump to fax detection if inside a macro |
ASTERISK-04267: [patch] [post 1.2] Asterisk as a Voice Mail Server for Other Softswitches |
ASTERISK-04268: [patch] TDD MODE ON command hangs up channel if TDD MODE not supported by that channel. |
ASTERISK-04269: [patch] Voicemail server/client MWI notifier + executing diaplan ext. upon IAX2/SIP registering/unregistering |
ASTERISK-04270: Asterisk connect to false proxy when it opens a call |
ASTERISK-04271: Context of peer is not set properly |
ASTERISK-04272: stuck SIP channels |
ASTERISK-04273: Zombies, MOH, and Bad Transfers |
ASTERISK-04274: [patch] Typo in chan_sip.c |
ASTERISK-04275: [patch] [post 1.2] improved loader |
ASTERISK-04276: [patch] add new function STRSTR returning position of substring in string |
ASTERISK-04277: [patch] rewritten chan_oss.c |
ASTERISK-04278: [patch] ExecIfTime kills your call when outside the timespec |
ASTERISK-04279: [patch] Allow for deletion and reparsing of global variables on extensions reload/pbx reload |
ASTERISK-04280: [PATCH] chan_sip.c bug in maddr handling in set_destination |
ASTERISK-04281: [patch] uninitialized variable in asterisk.c:main() |
ASTERISK-04282: [patch] wrong const in utils/astman.c |
ASTERISK-04283: [patch] If Curl() fails to fetch the URL it crashes asterisk |
ASTERISK-04284: Analog Modem transmission through Asterisk (fax included) |
ASTERISK-04285: [request] checking the result of Queue() |
ASTERISK-04286: [patch] H.323 compilation failure fix |
ASTERISK-04287: app_queue on more than one asterisk box |
ASTERISK-04288: [patch] New Function IFTIME |
ASTERISK-04289: [patch] Channel Reference Driver |
ASTERISK-04290: SetCIDNum: CID numbers including dots and spaces are incorrectly parsed by IAX peers |
ASTERISK-04291: [patch] ChanSpy ignores Zap/pseudo |
ASTERISK-04292: Zaptel 1.2.3 zaptel.init missing MODULES |
ASTERISK-04293: [patch] File ast_fileexists reports that file vm-nytt does not exist in Swedish in any format |
ASTERISK-04294: Patch associated with 0004350 causes very minor dependancy issue |
ASTERISK-04295: [patch] [needs testers] Add context and mailbox to database with odbc_storage |
ASTERISK-04296: Attended Transfer causes Crash |
ASTERISK-04297: Missing argument in pri_error |
ASTERISK-04298: bogus bound checks and useless code in chan_sip.c |
ASTERISK-04299: one bug and two useless blocks of code in chan_iax2.c |
ASTERISK-04300: [patch] explicit port number (5060) instead of DEFAULT_SIP_PORT |
ASTERISK-04301: [patch] wrong args to ast_copy_string in chan_sip.c:reply_digest() |
ASTERISK-04302: [patch] Asterisk crashes when forget the last ) in dial |
ASTERISK-04303: [patch] suggestd check on arguments to ast_add_extension*() |
ASTERISK-04304: Directory displays wrong language.. but play correct soundfiles |
ASTERISK-04305: [patch] Small fixes in show channeltypes |
ASTERISK-04306: Unable to complete attended transfer when running from startup script |
ASTERISK-04307: Guidelines for Doxygen documentation |
ASTERISK-04308: # transfers on SIP UA do not work after call is placed on hold |
ASTERISK-04309: Agent support ignores [featuremap] settings in features.conf |
ASTERISK-04310: [patch] hebrew say number |
ASTERISK-04311: [patch] changed app_addons_sql_mysql to set status variable instead of returning -1 |
ASTERISK-04312: [patch] ast_pbx_outgoing_app crashes if called will null appdata and sync > 0 |
ASTERISK-04313: SayUnixTime doesn't parse 21-29, 31 (twenty first, thirty first, etc) dates properly |
ASTERISK-04314: 1.0.8 Release Candidate |
ASTERISK-04315: Venezuela [VE] indications.conf |
ASTERISK-04316: [patch] Problem Compiling Add-ons On Mac OS X 10.4.1 |
ASTERISK-04317: [patch] new format AU |
ASTERISK-04318: [patch] Support for prompt playing in AGI WAIT_FOR_DIGIT cmd |
ASTERISK-04319: [patch] Support for prompt playing in AGI WAIT_FOR_DIGIT cmd |
ASTERISK-04320: Realtime+IAX and RSA auth |
ASTERISK-04321: internal call keeps ringing regardless that the external call has hung-up |
ASTERISK-04322: [patch] format_slin.c seek is broken |
ASTERISK-04323: ZapRAS does not work when used from Call File to place a call over Zaptel PRI |
ASTERISK-04324: Announcing Callers Position in Queue |
ASTERISK-04325: ${DIALSTATUS} broken |
ASTERISK-04326: [PATCH] Static function documentation |
ASTERISK-04327: [patch] UPGRADE.txt should note the change in argument format for the Record app |
ASTERISK-04328: [patch] Swedish indications.conf.sample |
ASTERISK-04329: [patch] Playtones to allow for midi notes as well as Hz freqs |
ASTERISK-04330: 2nd call with a Cisco Gateway produces One-Way-Audio |
ASTERISK-04331: [patch] WAV in res_features.c should be wav |
ASTERISK-04332: [patch] completed module.h documentation |
ASTERISK-04333: [patch] Improving the regexten functionality in SIP and IAX2 |
ASTERISK-04334: [patch] fix app_mp3 to work with http urls |
ASTERISK-04335: [patch] chan_sip.c: remove 4 copies of similar code. |
ASTERISK-04336: SIP Registration Inbound Call Sends to From Callerid |
ASTERISK-04337: [patch] chan_sip.c misc code cleanup |
ASTERISK-04338: possible bug in checking SIP authentication |
ASTERISK-04339: [patch] REGISTER "deadlock" between SPA's and Asterisk/Non-SPA Interoperability |
ASTERISK-04340: successful pin change on permission error |
ASTERISK-04341: Frame header allocation count drops below zero when using RECEIVE CHAR |
ASTERISK-04342: [patch] 'send text' CLI function in chan_alsa only sends one character. |
ASTERISK-04343: [patch] Small data types falsify results |
ASTERISK-04344: [patch] Make format_au compile on FreeBSD |
ASTERISK-04345: [patch] Replace "gave up" with "failed" |
ASTERISK-04346: [patch] Clean up transmit_invite |
ASTERISK-04347: [patch] Add conditional operator to the new expression parser |
ASTERISK-04348: [patch] Fixed some stuff in sounds.txt |
ASTERISK-04349: [patch] SIP registration possible crash |
ASTERISK-04350: Problem to pick up parked call |
ASTERISK-04351: [patch] Enhance "show channeltypes" with device state |
ASTERISK-04352: [patch] Add SIP domain support |
ASTERISK-04353: NI-2 PROGRESS 'Call is not end-to-end ISDN; further call progress information may be available inband' not passed across IAX? |
ASTERISK-04354: [patch] TDD MODE ON command hangs up channel if TDD MODE not supported by that channel. |
ASTERISK-04355: [patch] If no username/password is specified in cdr_odbc.conf, use NULL |
ASTERISK-04356: Fax detection will not properly jump to fax detection if inside a macro |
ASTERISK-04357: CVS-HEAD 2005-05-06 compilation error |
ASTERISK-04358: [patch] add new feature to app_math |
ASTERISK-04359: app_dial matches incorrect extension when exiting "Unable to create channel of type 'SIP'" |
ASTERISK-04360: Interop issues with Acme session border controller dropped calls after 30 seconds |
ASTERISK-04361: [patch] Format_au does not work on Solaris due to endian headers |
ASTERISK-04362: [patch] Unquoted string comparisons fail in expression parser |
ASTERISK-04363: [ASTCC] - Call Rating Issues - Patch Included |
ASTERISK-04364: [ASTCC] - Add Function - Calculate Charge from Web Page - Patch Included |
ASTERISK-04365: Documentation incorrect for new DB Functions |
ASTERISK-04366: [patch] new zonedata for de,ch,dk,cz,cn |
ASTERISK-04367: [patch] unused variables and middle-of-block declarations |
ASTERISK-04368: [patch] header order fixes for freebsd |
ASTERISK-04369: [patch] astmanproxy 1.0 - New Multithreaded Manager Proxy |
ASTERISK-04370: [patch] Change CODINGGUIDELINES to follow the guidelines in the examples |
ASTERISK-04371: [patch] Modify chan_iax2.c to use ast_strdupa correctly |
ASTERISK-04372: Queues stop calling members after a while |
ASTERISK-04373: [needs testing] Jukebox Music Manager AGI Script |
ASTERISK-04374: 407 Response Returned |
ASTERISK-04375: [Patch] check_expr -- a small program to check for possible probs in extensions.conf |
ASTERISK-04376: [patch] remove useless code |
ASTERISK-04377: possible minor bug in chan_iax2.c:calc_timestamp() |
ASTERISK-04378: [patch] remove extra call to ast_channel_walk_locked in ast_app_group_match_get_count |
ASTERISK-04379: [patch] Don't transfer to busy tone in a three-way call. (chan_zap.c) |
ASTERISK-04380: Asterisks sends responses to the incorrect port |
ASTERISK-04381: [patch] Fix for app_queue "Queues" manager action; output not properly terminated |
ASTERISK-04382: minor code cleanup in app_queue.c |
ASTERISK-04383: Call on hold gets dropped if the other phone goes on-hook within 1 second of pressing Recall. |
ASTERISK-04384: Using IAX2 encryption will result in segfault. |
ASTERISK-04385: Crash in app_queue.c |
ASTERISK-04386: [patch] Modify chan_iax2.c to use ast_strdupa correctly |
ASTERISK-04387: [patch] s extension is not used on incoming overlap call without extension |
ASTERISK-04388: [patch] replace hand-rolled timeval manipulation with generic functions. |
ASTERISK-04389: [patch] one-way-audio problem on multihomed host with bind to secondary address: |
ASTERISK-04390: ChanIsAvail not working with SIP |
ASTERISK-04391: Replace the config.guess in editline to a newer one. (supports, among others interix) |
ASTERISK-04392: [patch] revert ast_channel_walk_locked log message |
ASTERISK-04393: Build date omitted in version string on non-GNU systems |
ASTERISK-04394: [patch] variable declaration in the middle of a block |
ASTERISK-04395: [patch] Optional exit context and extension matching for app_background |
ASTERISK-04396: [patch] Change 405 to 501 for not implemented sip method |
ASTERISK-04397: added dtmfmode database option to mysql sipfriends |
ASTERISK-04398: [patch] Fix app_math compiling on <gcc3 |
ASTERISK-04399: [patch] man page for astgenkey, astman |
ASTERISK-04400: changing any iax peer name while using dnsmgr on reload will segfault. |
ASTERISK-04401: iax2 codec pref broken suspect recent create_addr changes in iax2 |
ASTERISK-04402: [patch] 2 more bugs in format_slin.c |
ASTERISK-04403: [patch] unable to german (de) and danish (dk) zone zones |
ASTERISK-04404: [request] Multiple lines from one SIP provider treated as one line in logs and CDR |
ASTERISK-04405: [patch] cdr.c: ast_cdr_getvar returns wrong answer/end time |
ASTERISK-04406: Asterisk doesn't compile after CVS checkout |
ASTERISK-04407: [patch] separate jitterbuffer debugging from iax debugging |
ASTERISK-04408: [patch] "cachetime" option for dundi.conf, making the hard coded cache time adjustable. |
ASTERISK-04409: [patch] Add function to read a text string from a channel. |
ASTERISK-04410: [patch] small patch to app_skel |
ASTERISK-04411: [patch] pciradio.c does not compile |
ASTERISK-04412: Monitor of two bridged Zap channels only records -in audio stream |
ASTERISK-04413: Local channels do not change to their connected trunk(Zap, SIP, ...) upon receiving Answer |
ASTERISK-04414: [patch] [post 1.2] Speex ultra wide band |
ASTERISK-04415: [patch] ResetCDR causes deadlock and/or segfault. |
ASTERISK-04416: [patch] FreeBSD portability and mpg123 termination bugs |
ASTERISK-04417: Uniden "workarounds" (nat=route) seems to be partly broken in CVS-HEAD |
ASTERISK-04418: [fixed-need testers] Crash with Audiocode UA set to DTMF notify |
ASTERISK-04419: app_dial (L) option seems to be broken |
ASTERISK-04420: Multiple INVITE's |
ASTERISK-04421: va_copy does not exist, res_config_odbc will not work |
ASTERISK-04422: [patch] Record "network provided number" as ANI and "user provided number" as Caller*ID |
ASTERISK-04423: CVS-HEAD to CVS-STABLE call problem |
ASTERISK-04424: [patch] add version files to pbx_ael.c |
ASTERISK-04425: documentation needed for (Try -DNO_CALIBRATION in Makefile) in the zaptel makefile |
ASTERISK-04426: iax2 indications not being followed |
ASTERISK-04427: Configurable Ringing Frequency and Voltage for Zaptel FXS ports from indications.conf?? |
ASTERISK-04428: Annex for g729 |
ASTERISK-04429: Tweak to the GSM Makefile to compile on the arm4vl |
ASTERISK-04430: Tweak to the GSM Makefile to compile on the arm4vl |
ASTERISK-04431: FastAGI / AGI command SET MUSIC ON class not working. |
ASTERISK-04432: Problem with utils.c causes seg fault |
ASTERISK-04433: [patch] add Callerid1 Callerid2 to manager Event: Link and Unlink |
ASTERISK-04434: Zaptel T1 sending double digits |
ASTERISK-04435: Increase TIMER_PERIOD_RING in chan_vpb |
ASTERISK-04436: Agent calls not recorded as agent-<agent-num> |
ASTERISK-04437: [patch] fix for channel.c timelimit bug |
ASTERISK-04438: [request] realtime multiple and failover database support |
ASTERISK-04439: [patch] 24h date format support on outgoing voicemail mails |
ASTERISK-04440: [patch] a classical deadlock in chan_h323 |
ASTERISK-04441: [patch] upgrade to check_expr to parse expressions, and give syntax errors if detected. |
ASTERISK-04442: Warning messages in cvs 1-0 stable |
ASTERISK-04443: Data Buffer Size Exceeded! leads to crash |
ASTERISK-04444: [patch] fix infinite loop in chan sip introduced from M4447 |
ASTERISK-04445: [patch] newjb - upon resync, delay of packet was inaccurately recorded |
ASTERISK-04446: a=silenceSupp:off - - - - |
ASTERISK-04447: [patch] missing \n on debug output |
ASTERISK-04448: sendDTMF inband does not send when a digit is repeated |
ASTERISK-04449: [patch] show config mappings (realtime stuff) |
ASTERISK-04450: [patch] Exten 500 broken in extensions.ael.sample |
ASTERISK-04451: [patch] Characters are removed from caller ID numbers that are not phone numbers |
ASTERISK-04452: [patch] Makefile fix for Solaris |
ASTERISK-04453: [patch] asterisk-addons does not use $(DESTDIR) for install |
ASTERISK-04454: [patch] Allow building asterisk for other CPUs than current system |
ASTERISK-04455: [patch] Pick up ANI (network provided number) from libpri |
ASTERISK-04456: possible confusion in configs/extensions.ael.sample |
ASTERISK-04457: Missing argument in q931.c causes asterisk to crash. Bug #4405 appears again |
ASTERISK-04458: pbx_gtkconsole does not compile after uncommenting the PBX_LIBS in the Makefile |
ASTERISK-04459: Calls completed and service level % figures are wrong |
ASTERISK-04460: [patch] Syntax of goto being used in extensions.ael.sample is broken in multiple locations. |
ASTERISK-04461: Patch from #4447 breaks auth with broadvoice |
ASTERISK-04462: [patch] Multiple syntax errors in examples listed in README.ael. |
ASTERISK-04463: [patch] Syntax errors in example of macro usage in README.ael. |
ASTERISK-04464: # in sip url |
ASTERISK-04465: Zap channel is not hung up when a DISCONNECT message with cause code 41 is gotten from PRI |
ASTERISK-04466: [patch] make install calls mkdir -p $(DESTDIR)$(ASTSBINDIR) twice |
ASTERISK-04467: [branch] setting the musicclass doesn't work as one might expect (hold) |
ASTERISK-04468: [patch] [post 1.2] channel.c ast_waitfor*() cleanup/speedup |
ASTERISK-04469: [patch] Remove locks in usecount() in formats |
ASTERISK-04470: [patch] cdr_addon_mysql crashes under heavy use |
ASTERISK-04471: SayAlpha does not say "9" as a digit (from say.c) |
ASTERISK-04472: [patch] chan_sip crash w/ hitachi cable 5000 phone & transfer |
ASTERISK-04473: [patch] ZapRAS "destroys" channel when done |
ASTERISK-04474: [patch] res_agi.c, launch_netscript() does not handle any system call interuption |
ASTERISK-04475: [patch] ast_recvchar() ast_recvtext() fix memory leak and bug |
ASTERISK-04476: [patch] ODBC voicemail message storage with stutter dial and light alerts |
ASTERISK-04477: [change request] swap arguments to ast_tvdiff_ms() |
ASTERISK-04478: [patch] Manager Hold events are only generated for bridged channels |
ASTERISK-04479: [patch] externnotify clairfications |
ASTERISK-04480: H.323 context |
ASTERISK-04481: AGI GET VARIABLE not finding variables |
ASTERISK-04482: AGI DATABASE GET not working |
ASTERISK-04483: Dialplan timeout problem |
ASTERISK-04484: [patch] wrong argument to ast_copy_string |
ASTERISK-04485: [patch] simplified string handling in cdr_pgsql and cdr_odbc |
ASTERISK-04486: password not changed immidietly with externpass |
ASTERISK-04487: [patch] simplify macros for inline/not inline API functions |
ASTERISK-04488: [patch] In some circumstances, we should not say "o'clock" |
ASTERISK-04489: [patch] code cleanup |
ASTERISK-04490: [patch] RECEIVE TEXT function hangs up channel if timeout times out |
ASTERISK-04491: [patch] speex dtx |
ASTERISK-04492: Still timelimit bug with h323 channel driver !! |
ASTERISK-04493: Asterisk behind a Cisco PIX 515E firewall using fixup protocol |
ASTERISK-04494: [request] Add Fax detection to SIP (and potentially other channels) |
ASTERISK-04495: [request] Add E option to Dial() cmd to disable echo cancellation |
ASTERISK-04496: Entering meetme with pin access, hangup at pin entry meetme exists with 0 users |
ASTERISK-04497: Stale nonce with multiline registration. |
ASTERISK-04498: [patch] ast_recvtext function assumes text received is null terminated |
ASTERISK-04499: ${AGENTBYCALLERID_${CALLERID} variable is not updated when kicked |
ASTERISK-04500: someone can place an unauthenticated call even though auth=rsa is defined on the host |
ASTERISK-04501: 0004343: REGISTER "deadlock" between SPA's and Asterisk/Non-SPA Interoperability |
ASTERISK-04502: [patch] dir is not always necessary |
ASTERISK-04503: original spam.gsm missing from asterisk-sounds |
ASTERISK-04504: [patch] segfault when Asterisk auto-rotates log on getting a SIGXFSV on a log file write (which means file too large) |
ASTERISK-04505: [patch] (for channel.c) At monitor command in -out file written only noise when used chan-misdn. |
ASTERISK-04506: [patch] Update editline to compile on cygwin |
ASTERISK-04507: Asterisk Secfault |
ASTERISK-04508: Another Asterisk crash |
ASTERISK-04509: Little patch to allow incrementing of RetryTime in outgoing spool callfiles |
ASTERISK-04510: Add option to Playback() to allow user to skip the message with any DTMF |
ASTERISK-04511: DTMF Registering in IAX Debug output but ignored for IVR purposes |
ASTERISK-04512: [patch] 'send text' CLI command terminates sent text with space instead of newline |
ASTERISK-04513: When Primary D-Channel goes down q931_hangup may receive a wrong (q931_call *) argument |
ASTERISK-04514: [patch] after running for a while, chan_sip permanently stops registration attempts on the first failure |
ASTERISK-04515: Realtime SIP update_peer in chan_sip.c |
ASTERISK-04516: [request] Asterisk ignores/caches phone number/username and context for Broadvoice SIP trunks |
ASTERISK-04517: [patch] #include "directory" causes asterisk hang with 100%cpu |
ASTERISK-04518: Indications.conf has two typos |
ASTERISK-04519: Zaptel compiles but fails to load with 2.6.13 series kernels |
ASTERISK-04520: [patch] "autosupport" script not recognized as such |
ASTERISK-04521: file.c: ast_readfile deletes files |
ASTERISK-04522: [patch] man pages for autosupport and safe_asterisk |
ASTERISK-04523: [patch] "autosupport" script not recognized as such |
ASTERISK-04524: [patch] inc abandons when the call leaves the queue from failure/issue. |
ASTERISK-04525: [patch] man pages for autosupport and safe_asterisk |
ASTERISK-04526: [patch] segfault when Asterisk auto-rotates log on getting a SIGXFSV on a log file write (which means file too large) |
ASTERISK-04527: Private structure not found in progress. |
ASTERISK-04528: [patch] ZapRAS "destroys" channel when done |
ASTERISK-04529: [patch] Make webvmail is not very adaptable to varying apache configurations |
ASTERISK-04530: ZapRAS leaves PRI channel in "unclean" state |
ASTERISK-04531: app_dial.c:362 wait_for_answer: Unable to forward frame |
ASTERISK-04532: [patch] Cannot compile on Freebsd after introduction of strndup() in channel.c |
ASTERISK-04533: Limit fails to hangup call when no RTP |
ASTERISK-04534: [patch] correct plural 's' for serveral CLI outputs |
ASTERISK-04535: [patch] use the mailbox syntax in password file for app_disa |
ASTERISK-04536: Entering meetme with pin access, hangup at pin entry meetme exists with 0 users |
ASTERISK-04537: [request] RDNIS not working properly - Reports LAST redirect instead of the ORIGINAL redirect |
ASTERISK-04538: Capi |
ASTERISK-04539: [patch] RTP rfc2833 out of sequence problem |
ASTERISK-04540: [patch] if-else statements in extensions.ael does not work as documented |
ASTERISK-04541: [patch] Added the ability to 'escape' durning recording and review message. |
ASTERISK-04542: Incoming Zaptel calls to Polycom 600 hang Asterisk |
ASTERISK-04543: Cygwin Patches (additional files needed for functions not in cygwin) |
ASTERISK-04544: Cygwin patches (CDR Subdir) |
ASTERISK-04545: chan_skinny crashes asterisk when the skinny phone recieved a call. |
ASTERISK-04546: Cygwin Patches (apps dir) |
ASTERISK-04547: [request] Add 'UserField:' parameter to pbx_spool.c |
ASTERISK-04548: Cygwin Patches (res dir) |
ASTERISK-04549: [patch] remove libstrfunc dependency |
ASTERISK-04550: [patch] simplify ui string handling in pbx.c |
ASTERISK-04551: [patch] misc code simplifications in pbx.c |
ASTERISK-04552: [change request] be careful with printf format specifiers. |
ASTERISK-04553: Cygwin Patches (channels dir) |
ASTERISK-04554: Placing SIP -> PRI call on hold more then 4 times drops call |
ASTERISK-04555: Cygwin Patches (core\includes\root dir) |
ASTERISK-04556: ASTCC generates error while compiling.. |
ASTERISK-04557: [patch] Customizable periodic announcement for app_queue |
ASTERISK-04558: [patch] Cygwin portability |
ASTERISK-04559: manager.c spewing "accept_thread: Accept returned -1: Resource temporarily unavailable" |
ASTERISK-04560: Voicemail doesn't detect hangup prior to starting recording |
ASTERISK-04561: [patch] variable declaration in the middle of a block |
ASTERISK-04562: [patch] use of uninitialized variable |
ASTERISK-04563: [patch] fixes to vm_intro_it() |
ASTERISK-04564: channel_find_locked( NULL, mysearchstring, 0) works not as proposed |
ASTERISK-04565: Asterisk won't build if LOW_MEMORY defined |
ASTERISK-04566: [patch] callgroup/pickupgroup not parsed correctly |
ASTERISK-04567: Enabling rtcachefriends prevents phones on different servers from calling each other |
ASTERISK-04568: MOH crashes. |
ASTERISK-04569: [patch] remove asterisk.conf from being loaded twice. |
ASTERISK-04570: [patch] ACF QUEUEAGENTCOUNT() and multi-digit exits |
ASTERISK-04571: "file.c: No such format" causes Asterisk to crash |
ASTERISK-04572: accept_thread INFINITE LOOP |
ASTERISK-04573: [patch] Allow restart char for ast_control_streamfile |
ASTERISK-04574: [patch] Manager action "QueueStatus" does not send "StatusCompleteEvent" when done |
ASTERISK-04575: no colors on console when asterisk is is running as remote console (asterisk -rc) |
ASTERISK-04576: [patch] No ringback on SIP calls after answer |
ASTERISK-04577: Typo in "show application playtones" |
ASTERISK-04578: [patch] Function to retrieve SIP channel information |
ASTERISK-04579: Asterisk unilaterally CANCELs a Dial |
ASTERISK-04580: Problem interpreting FACILITY during setup from HiPath 4k |
ASTERISK-04581: [patch] Daemon for zaptel status |
ASTERISK-04582: ${MACRO_EXTEN} is not always |
ASTERISK-04583: [patch] make use of ast_sched_dump and add missing Makefile directives |
ASTERISK-04584: bug id 752 turns on call recording by defult abort calls if no sound files |
ASTERISK-04585: [patch] The flag "transfertobusy" doesn't work. |
ASTERISK-04586: [patch] ast_translate doesn't advance outgoing timestamps during silent periods |
ASTERISK-04587: crashes with core dump in my_strcasecmp_8bit from /usr/lib/libmyodbc3.so |
ASTERISK-04588: USE_ODBC_STORAGE broken in head |
ASTERISK-04589: ASTCC its cutting the numbers |
ASTERISK-04590: [patch] REMOTE_ADDR Address Exposure to Dialplan |
ASTERISK-04591: [patch] RTP rfc2833 out of sequence problem |
ASTERISK-04592: __builtin_expect in libpri:copy_string.c prevents compilation under gcc 2.95.3 |
ASTERISK-04593: [patch] Man pages for some zaptel commands |
ASTERISK-04594: [patch] added undef for define format |
ASTERISK-04595: [patch] Changes %i to %d |
ASTERISK-04596: Asterisk locks when ChanIsAvailable used with an IAX channel & SIP channel |
ASTERISK-04597: Patch for french |
ASTERISK-04598: [patch] Signal handling on Solaris loses handler after one signal |
ASTERISK-04599: Solaris res_agi failed to find __ast_malloc symbol |
ASTERISK-04600: agent making outgoing call after logging in using agentcallbacklogin is reported as unavailable instead of busy |
ASTERISK-04601: [patch] Manager action "Agents" does not send ActionID back, does not send an ACK and is missing an AgentsCompleteEvent |
ASTERISK-04602: [patch] + new files Fix ChanSpy |
ASTERISK-04603: [request] ENUM lookup functionality extension |
ASTERISK-04604: [patch] app_forkcdr.c bug fix |
ASTERISK-04605: [patch] Manager action "DBGet" does not send ActionID back and is missing locking |
ASTERISK-04606: [new_dialplan_function] SIPPEER function |
ASTERISK-04607: [patch] retrieve SIP To: header for ISDN legacy use |
ASTERISK-04608: tvfix negative timestamps |
ASTERISK-04609: iax predicted timestamp skew |
ASTERISK-04610: [patch] Logger config should tell people to keep debug mode off |
ASTERISK-04611: [patch] Devicestate notification in clear text |
ASTERISK-04612: 'restart' fails with asterisk -U |
ASTERISK-04613: [new app] app_muxmon |
ASTERISK-04614: [patch] Formatting pbx.c |
ASTERISK-04615: [patch] Addition of user definable ExitStatus field to manager Hangup event. |
ASTERISK-04616: [patch] control_stream_file (like ControlPlayback) for AGI |
ASTERISK-04617: [patch] Formatting io.c |
ASTERISK-04618: move strtoq to utils.c and ifdef using HAVE_STROQ |
ASTERISK-04619: [patch] alignment/truncation of 'show channels' output and new fields callerid, bridge, duration |
ASTERISK-04620: [patch] CODING-GUIDELINES: tabsize 4, spaces for alignment of in-line comments, explanation of indent parameters |
ASTERISK-04621: [patch] DISA should not set accountcode if blank |
ASTERISK-04622: [patch] Fix struct timeval portability |
ASTERISK-04623: ${PRIORITY} is not accessible with AGI |
ASTERISK-04624: [patch] ast_tvdiff_ms is buggy due to different truncation effect of gcc for negative and positive numbers |
ASTERISK-04625: [patch] Speex preprocessor + fixed-point compatibility etc |
ASTERISK-04626: chan_iax2 sending crazy timestamps that completely freak remote jitter buffer |
ASTERISK-04627: VoiceMail "change Password" option is invalid |
ASTERISK-04628: asterisk freeze if mySQL server unreachable |
ASTERISK-04629: [patch] Formatting acl.c |
ASTERISK-04630: [patch] DISA should not set accountcode if blank |
ASTERISK-04631: [patch] Formatting db.c |
ASTERISK-04632: [patch] Formatting astmm.c |
ASTERISK-04633: asterisk cannot be run by a non-root user |
ASTERISK-04634: [patch] segfault on using SIP_HEADER on dead channel |
ASTERISK-04635: [patch] Format of chan_sip |
ASTERISK-04636: [patch] info about asterisk.conf |
ASTERISK-04637: [patch] queue members with multiple devices. |
ASTERISK-04638: [patch] add a feature_timer member to bridge_config |
ASTERISK-04639: IAX2 one way audio with jitterbuffer = on |
ASTERISK-04640: [patch] Formatting, small code change |
ASTERISK-04641: Improved comment for offset tweaking in p->offset that kpfleming asked for in #4747 |
ASTERISK-04642: [patch] app_authenticate: hide real passwords from CDR (account code) |
ASTERISK-04643: [patch] Change format of "show version file" |
ASTERISK-04644: [patch] Manager action "AgentCallbackLogin" and "AgentLogoff" added |
ASTERISK-04645: [patch] for "Dropping voice to exceptionally long queue" issues |
ASTERISK-04646: [patch] alignment mod to blank line after "show agents" output |
ASTERISK-04647: [patch] Format of plc.c |
ASTERISK-04648: [patch] Format of loader.c |
ASTERISK-04649: [patch] chan_zap sets the numcomplete parameter to pri_sr_set_called backwards , so our numbers are never complete... |
ASTERISK-04650: chan_sip2.c unable to compile |
ASTERISK-04651: Outbound Proxy Support |
ASTERISK-04652: 0002859: [patch] Outbound proxy support |
ASTERISK-04653: Callscreen function in app_dial not working |
ASTERISK-04654: include file in addons perl scripts |
ASTERISK-04655: [patch] Small addition to README.MP3 |
ASTERISK-04656: [patch] Fixing format of README.jitterbuffer |
ASTERISK-04657: [patch] Update to README.extconfig and README.realtime |
ASTERISK-04658: [patch] Formatting fixes for res_features |
ASTERISK-04659: [patch] added Russian syntax (ast_say_number_full_ru) |
ASTERISK-04660: [patch] Formatting fixes for channel.c/channel.h |
ASTERISK-04661: [patch] Adding cleartext hangupcause to BYE and CANCEL |
ASTERISK-04662: [patch] Bugfix: Only mention port in URI if it's not 5060 |
ASTERISK-04663: Macro and Realtime Extensions |
ASTERISK-04664: [patch] Formatting for chan_local |
ASTERISK-04665: [patch] Formatting for chan_features |
ASTERISK-04666: [patch] Add ENUMSTATUS result variable to enumlookup |
ASTERISK-04667: Address family missing in chan_sip.c |
ASTERISK-04668: Asterisk voicemail odbc storage |
ASTERISK-04669: Realtime SIP entries show UNKNOWN status in show sip peers |
ASTERISK-04670: [patch] RTP.c comments, log messages |
ASTERISK-04671: Cygwin portability patch for Asterisk 1.0.9 |
ASTERISK-04672: [patch] code cleanup - chan_sip.c has two almost identical functions 'get_in_brackets' and 'ditch_braces' |
ASTERISK-04673: [patch] code cleanup - chan_sip.c use of 'ast_buildstring' and wrapper for adding Content-Length to SIP replys |
ASTERISK-04674: [patch] [post 1.2] whitespace cleanup for chan_sip.c according to CODING GUIDELINES |
ASTERISK-04675: Error from get_in_brackets from chan_sip 1.789, truncated line? |
ASTERISK-04676: Asterisk doesn't signalize end of hold |
ASTERISK-04677: [patch] Events manager command with EventMask set to 'on' does not return anything |
ASTERISK-04678: [patch] Enter wrong voicemail password will make Asterisk segfault |
ASTERISK-04679: [patch] MWI compliance with RFC3842 - "Message-Account" in NOTIFY header |
ASTERISK-04680: [patch] Compiler errors on chan_modem |
ASTERISK-04681: [patch] Get back pridump into working stage |
ASTERISK-04682: [patch] Enhancement for ztmonitor |
ASTERISK-04683: [request] Change the menu for mailbox options so it does not break 1.0 compatiblity. |
ASTERISK-04684: [patch] New CLI: SIP show settings |
ASTERISK-04685: [patch] app_authenticate.c:123: warning: `md5secret' might be used uninitialized in this function |
ASTERISK-04686: [patch] Fix SENDTEXTSTATUS for app_sendtext |
ASTERISK-04687: AEL: "default" case in switch statement does not appear operational |
ASTERISK-04688: [patch] TRANSFERSTATUS for app_transfer |
ASTERISK-04689: [patch] Addition to README.asterisk.conf |
ASTERISK-04690: [patch] Add SENDURLSTATUS for app sendurl() |
ASTERISK-04691: template matching code needs to be case sensitive |
ASTERISK-04692: manager command queuepause should not be case sensitive |
ASTERISK-04693: manager command queuepause should not be case sensitive |
ASTERISK-04694: [patch] Enhance the dialplan function SIP_PEER |
ASTERISK-04695: template matching code needs to be case sensitive |
ASTERISK-04696: [patch] app_rand |
ASTERISK-04697: Multiple compile warnings under GCC 4 |
ASTERISK-04698: AEL parsing IF-Else incorrectly |
ASTERISK-04699: [branch] IPv6 support in chan_[sip,iax2] |
ASTERISK-04700: [patch] chan_sip.c: qop=auth syntax error |
ASTERISK-04701: [patch] Mec2 Timed Agressive Cancellation |
ASTERISK-04702: CLI prompt always shown underneath cli command output |
ASTERISK-04703: [patch][post 1.4] New codec negotiation algorithm |
ASTERISK-04704: [patch] astrisk crashes with zaptel-pri and overlapdial=yes |
ASTERISK-04705: Disable "!" from CLI using features.conf or any other .conf |
ASTERISK-04706: [patch] show function is now not case-sensitive |
ASTERISK-04707: [Patch] Make Pager subject and pager body customizable |
ASTERISK-04708: [patch] add busypattern= to zapata.conf so end-of-call busydetector can know and detect actual busy tone lengths |
ASTERISK-04709: app_chanspy |
ASTERISK-04710: [patch] Make Realtime SIP Clusterable |
ASTERISK-04711: dtmf inband with g729 |
ASTERISK-04712: 'n' option on cmd Queue is not currently functional |
ASTERISK-04713: [patch] Allow DTMF to be logged |
ASTERISK-04714: GROUP_COUNT crashes if category specified and group is empty |
ASTERISK-04715: [patch] AEL seems to have a maximum character limit |
ASTERISK-04716: [patch] [post 1.2] present SIP Call-ID of ringing call in NOTIFYs to subscribed phones |
ASTERISK-04717: LEN() function seems to segfault asterisk when used with big variables |
ASTERISK-04718: [request] user profiles & remote access |
ASTERISK-04719: [patch] chan_sip.c: qop=auth syntax error |
ASTERISK-04720: [patch] Memory Leak(tm) in res_musiconhold.c |
ASTERISK-04721: [patch] Memory Leak(tm) in app_voicemail.c |
ASTERISK-04722: Redhat init script does not detect an already running instance |
ASTERISK-04723: [patch] [post 1.2] make the 'RECORD' button on the snom phones work |
ASTERISK-04724: asterisk -rx not exit on linuxppc |
ASTERISK-04725: asterisk -rx not exit on linuxppc |
ASTERISK-04726: bug with AMI - Originate |
ASTERISK-04727: bug with AMI - Originate |
ASTERISK-04728: tos tag broken in iax.conf |
ASTERISK-04729: Problems with the # character |
ASTERISK-04730: [PATCH] show cid.cid_num and cid.cid_name in meetme list |
ASTERISK-04731: Digital Receptionist error |
ASTERISK-04732: [patch] ISQUEUEMEMBER([queuename],channel) |
ASTERISK-04733: [patch] allows asterisk to send SIGHUP to child AGI scripts on hangup |
ASTERISK-04734: [patch] allow use of custom asterisk functions from AGI |
ASTERISK-04735: [patch] 'iax2 show channels' causes immediate segfault |
ASTERISK-04736: [patch] Functions for url-encoding and decoding |
ASTERISK-04737: [patch] Func_tonezone: COUNTRY |
ASTERISK-04738: [patch] Additional sounds |
ASTERISK-04739: [patch] option "a" of application app_authenticate stop to work with the MD5 implementation |
ASTERISK-04740: [patch] Don't ignore ACK on INVITE |
ASTERISK-04741: An wrong authname in Invite for 407 proxy authen |
ASTERISK-04742: [patch] Update README.asterisk.conf |
ASTERISK-04743: [patch] Directed Call Pickup Support |
ASTERISK-04744: [patch] chan_h323.c fails to compile |
ASTERISK-04745: Remove /contrib/scripts/addmailbox |
ASTERISK-04746: Remote /contrib/scripts/iax-friends.sql |
ASTERISK-04747: Remove /contrib/scripts/sip-friends.sql |
ASTERISK-04748: format_mp3 Music on hold causes segfault |
ASTERISK-04749: [patch] [post 1.2] Pass AST_CONFIG_DIR to AGI scripts as either env or in variable exchange |
ASTERISK-04750: Multiple Agents cannot log into AgentCallBackLogin at the same time. |
ASTERISK-04751: SIP: Maximum retries exceeded on call 00120138-3ef9019f-27a0903e-7062e8d4 |
ASTERISK-04752: Originate channel Agent ties up agent (Agentcallbacklogin) |
ASTERISK-04753: [PATCH] AEL builds statement that won't execute on if with : operator using character classes |
ASTERISK-04754: Segfault when doing tranfers with a SNOM phone |
ASTERISK-04755: [patch] Identify Asterisk "urgent handler" message |
ASTERISK-04756: All sounds are stopped since 1.0.7-stable |
ASTERISK-04757: [patch] [post 1.2] support for using priority labels and dialplan functions for asterisk manager operations |
ASTERISK-04758: Voicemail hangs up after 1 min when calling from other asterisk server |
ASTERISK-04759: vasprintf in utils.c does not implement correct semantics on Solaris |
ASTERISK-04760: priority "a" in macro not recognized while voicemail app is playing "digits" of an extension |
ASTERISK-04761: [patch] [post 1.2] app_roundrobin - IAX load balancing |
ASTERISK-04762: seg fault when hanging up during app_voicemail vm_authenticate |
ASTERISK-04763: sip.conf Codec Order Not Recognized |
ASTERISK-04764: crash with missing extension? |
ASTERISK-04765: crash with missing extension? |
ASTERISK-04766: [patch] adding further compressed SIP aliases |
ASTERISK-04767: Previous patch for bug #4882 (Solaris vasprintf issues) freed a pointer before use |
ASTERISK-04768: One way audio betwen 1.0 and HEAD with trunktimestamps=no |
ASTERISK-04769: [patch] Add functions CUT and SORT |
ASTERISK-04770: Asterisk Crashes under heavy sip load |
ASTERISK-04771: Documentation does not match use - Festival interface. |
ASTERISK-04772: Dynamic codecs (eg. 97 - ilbc) are not handled correctly when asterisk steps out of the media path (canreinvite=yes) |
ASTERISK-04773: SIPAddHeader application has wrong channeltype check |
ASTERISK-04774: [patch] Record application never returns if no input is received from channel |
ASTERISK-04775: [patch] strings.h will not compile on Solaris due to vasprintf prototype declaration |
ASTERISK-04776: [patch] utils/Makefile refers to ../mkdep instead of ../build_tools/mkdep |
ASTERISK-04777: [patch] Add standard locations for Solaris unbundled software installation to main Makefile |
ASTERISK-04778: [patch] SIP over TCP project |
ASTERISK-04779: [patch] SIP over TCP project |
ASTERISK-04780: One way audio when doing IAX2 transfer |
ASTERISK-04781: Setting CallerID has no effect if doing SIP-SIP call |
ASTERISK-04782: origination through manager with empty variable field causes crash |
ASTERISK-04783: [patch] new musiconhold.conf parser, allow source format to be other than signed linear |
ASTERISK-04784: [patch] [post 1.2] Added support to cdr_pgsql.c to spool queries when lost database connection |
ASTERISK-04785: mistagging all queue completes as abandoned |
ASTERISK-04786: [patch] Allow calling Set with a function that uses | delimeters |
ASTERISK-04787: ExtensionState returns 0 when a zap connected extension is off hook and on a call |
ASTERISK-04788: Fix for bug 4771 stops overlapdial on PRI E1 as empty/partial number is sent as complete |
ASTERISK-04789: tor2 doesn't work on x86_64 |
ASTERISK-04790: [patch] Pointer (& data) corrupted on exit from get_input() in manager.c whenever poll() fails |
ASTERISK-04791: Inconsistent use of User/Username attribute in manager event "Registry" |
ASTERISK-04792: sip reload breaks dtmf via rfc2833 |
ASTERISK-04793: robotic audio in sip and voicemail + dbl speed vm |
ASTERISK-04794: [patch] [post 1.2] peer or user match priority option |
ASTERISK-04795: bad audio and lots of errors with speex |
ASTERISK-04796: zaptel install |
ASTERISK-04797: ToS can't be set properly |
ASTERISK-04798: Calls dropped on blind transfer using IAX2 |
ASTERISK-04799: In ATXFER, when the transferrer hangs up, the transferee can dial *2 to do a new transfer |
ASTERISK-04800: RTCP and NAT issue |
ASTERISK-04801: [patch] tos tag broken in iax.conf, fix for cvs fix |
ASTERISK-04802: [patch] enable dnsmgr at startup |
ASTERISK-04803: Global variables not evaluated by Realtime in extension name |
ASTERISK-04804: Solaris mpg123 doesn't die properly for musiconhold |
ASTERISK-04805: [patch] Remove realtime priority for AGI scripts |
ASTERISK-04806: Can not connect with asterisk -r when asterisk is running as non-root |
ASTERISK-04807: [patch] Recent changes to acl.c cause Solaris compile failure |
ASTERISK-04808: re-open of #4418 |
ASTERISK-04809: [request] Make it possible to retrieve SIP headers on a REFER (SIP transfer) |
ASTERISK-04810: [patch] [post-1.2] sendDTMF(f) doesn't flash the switch hook |
ASTERISK-04811: [patch] [post 1.2] react to call progress tones |
ASTERISK-04812: [patch] [post 1.2] 0 and * dont work right |
ASTERISK-04813: [patch] event_log = no causes asterisk to not start |
ASTERISK-04814: [patch] indications for [es] |
ASTERISK-04815: [request] [post 1.2] dialplan order / match order problems |
ASTERISK-04816: [patch] Don't dump core at re-registration |
ASTERISK-04817: Asterisk doesn't detect a device disconnect while in a call |
ASTERISK-04818: bridged call with FXO port not translating DTMF tones into pulsedial when pulsedial=yes |
ASTERISK-04819: ENUM Lookup REGEXP limited to 80 Characters |
ASTERISK-04820: Asterisk does not handle ringing on ISDN calls when PROGRESS sent before ALERT |
ASTERISK-04821: [patch] Format string error in cli.c |
ASTERISK-04822: Add sound fiile valid.gsm |
ASTERISK-04823: musiconhold leaving zombie mpg123 processes behind |
ASTERISK-04824: [PATCH] Incorrect copy and paste into receive text AGI Command |
ASTERISK-04825: [patch] [post 1.2] ENUM Lookup for non SIP/IAX2/TEL... |
ASTERISK-04826: ooH323 from asterisk-addons compile broken on x86_64 |
ASTERISK-04827: [patch] acl.c IPTOS_MIN |
ASTERISK-04828: The script contrib/safe_asterisk relies on asterisk in $PATH |
ASTERISK-04829: [patch] cdr_addon_mysql - Lost cdr records, hanging server, spooling sql statements |
ASTERISK-04830: [patch] [post 1.2] MeetMe - Added support for a conference max time |
ASTERISK-04831: Calling a macro before calling a dial command causes billsec and duration to be the same and also reflect incorrect elapsed time |
ASTERISK-04832: [patch] app_voicemail - We should find the next message number before we play the beep. |
ASTERISK-04833: [patch] iLBC_decode/encode uses memcpy where it should use memmove because the memory might overlap. |
ASTERISK-04834: [patch] init matchstr in cli.c |
ASTERISK-04835: [patch] memset dnsstate before use. |
ASTERISK-04836: tvfix too large timestamp |
ASTERISK-04837: now variable is used unintialized when 'n' option is set |
ASTERISK-04838: [patch] Misspelling: Lauching (Didn't your mothers tell you to stop "Lauching"?) |
ASTERISK-04839: [patch] Not correct description text for IF function. |
ASTERISK-04840: chan_h323 crashes when gatekeeper used and calls addressed with IP address |
ASTERISK-04841: [post 1.2] patch for easier cross-compiling (and also for realtime preemptive patched kernels) |
ASTERISK-04842: SIP 404 returns ${DIALSTATUS}=CONGESTION |
ASTERISK-04843: [patch] astmm fails to compile |
ASTERISK-04844: [post 1.2] Cross-compiling adjustments |
ASTERISK-04845: [patch] Added support for custom URI options in INVITE |
ASTERISK-04846: [patch] res_config_mysql ignores database specified in extconfig.conf |
ASTERISK-04847: crash with iax2 debug |
ASTERISK-04848: [request] building Zaptel requires full kernel source install & make |
ASTERISK-04849: [patch] voicemail password change crash |
ASTERISK-04850: chan_h323 crashes asterisk when send calls to an ip and is registered to a gatekeeper |
ASTERISK-04851: MailboxExists doesn't seem to support 'n' priorities |
ASTERISK-04852: [patch] a little bit gooder english on d-chan |
ASTERISK-04853: IAX2Provision fails - No PROVISION frame sent |
ASTERISK-04854: After upgrading to echo can card kernel panics and popping/mute with digit sounds. |
ASTERISK-04855: 403 Forbidden in REGISTER violates rfc3261 |
ASTERISK-04856: [patch] [post 1.2] Japanese SayNumber functionality |
ASTERISK-04857: [patch] when in pri network mode, q.931 always represents the channels as preferred instead of exclusive, causing call drops |
ASTERISK-04858: Called party of outbound calls cannot enter DTMF |
ASTERISK-04859: app_meetme+chan_sip retries RTP packets indifinetly |
ASTERISK-04860: core dump when i try to authenticate with realtime mysql |
ASTERISK-04861: [patch] [post 1.2] prefix for UNIQUEID |
ASTERISK-04862: [patch] Failure to send Out of Band DTMF (rfc2833) key indications via h.323 - will receive, but not send |
ASTERISK-04863: [patch] like clause not case sensitive in show modules |
ASTERISK-04864: [patch] counter in show functions |
ASTERISK-04865: [patch] Counter in show version files |
ASTERISK-04866: RTP error seems to prevent DTMF from being sent |
ASTERISK-04867: [patch] handle BYE on non-answered call correctly |
ASTERISK-04868: [patch] SIP INVITE with bad auth data results in dialplan hang |
ASTERISK-04869: [patch] [post 1.2] wrappers for memory allocation error handling |
ASTERISK-04870: [patch] don't require source to be built on 'make samples' |
ASTERISK-04871: External Incoming CDRs not logged in database |
ASTERISK-04872: [patch] A potential solution to PauseQueueMember / UnPause issues. |
ASTERISK-04873: [patch] libpri/Makefile has quite a few problems when used on a Solaris system |
ASTERISK-04874: enabling sip debugging, and using 'sip reload' from cmd disables debug |
ASTERISK-04875: [patch] Counter in show file formats |
ASTERISK-04876: [patch] Counter in show keys |
ASTERISK-04877: [patch] Make streamplayer compile on FreeBSD |
ASTERISK-04878: [patch] res_features fixup |
ASTERISK-04879: [patch] show vmexten in "sip show settings" |
ASTERISK-04880: [post 1.2] 183 handling |
ASTERISK-04881: chan_sip2.c |
ASTERISK-04882: [patch] Freebsd fix for streamplayer |
ASTERISK-04883: [patch] [post 1.2] some improvement in callprogress=yes code in dsp.c |
ASTERISK-04884: Asterisk Calls VoiceXML Application Deployed on SipXVxml -- in Asterisk 1.0.9 -- chan_sip.c |
ASTERISK-04885: Problems with forward when busy on using sip phones |
ASTERISK-04886: [patch] Cannot jump to an n(label) priority |
ASTERISK-04887: [patch] implement call pickup on Snom phones |
ASTERISK-04888: [patch] [post 1.2] cdr_csv logging parameters in cdr.conf |
ASTERISK-04889: manager event reporting cannot be dynamically set |
ASTERISK-04890: [patch] Add timer T1 support for NOTIFY messages |
ASTERISK-04891: [patch] Remove some CLI messages (and RTP de-allocation) |
ASTERISK-04892: [patch] utils/streamplayer.c compile problem on openbsd |
ASTERISK-04893: queue_log simultaneous write problem |
ASTERISK-04894: [patch] Add docs for recent changes to qualify= |
ASTERISK-04895: [patch] AEL does not properly parse Apps with equals in the arguments |
ASTERISK-04896: expires header parameter should be matched case-insensitively |
ASTERISK-04897: [patch] [post 1.2] Show manager permissions |
ASTERISK-04898: [patch] Disable musiconhold if we have no music |
ASTERISK-04899: [patch] Adds documentation to srv.h |
ASTERISK-04900: [patch] Cleaning up doxygen docs for logger.h |
ASTERISK-04901: [patch] Update extensions.txt |
ASTERISK-04902: [patch] Update manager.h |
ASTERISK-04903: Agent still logedin when direct extension UNREACHABLE. |
ASTERISK-04904: [request] [post 1.2] Feature to allow for agents to be logged off automatically after missing a number of calls |
ASTERISK-04905: When doing an attended transfer from a queue using the fatues-based transfer, Asterisk cashes |
ASTERISK-04906: [patch] [post 1.2] Allow multiple headers in UserEvents |
ASTERISK-04907: [patch] Remove extra ast_mutex_unlock() in chan_vpb.c |
ASTERISK-04908: [patch] Fix comments on masq |
ASTERISK-04909: core dump with astmm |
ASTERISK-04910: Call wasn't interrupted when Dial() was used with S() option. |
ASTERISK-04911: for backtraces |
ASTERISK-04912: [patch] rport stripped from all via headers not just top one |
ASTERISK-04913: Voicemail with MySQL won't find mailbox if db connection is lost |
ASTERISK-04914: CLI tab completion works with partial command followed by <space><tab> but partial commands do not. |
ASTERISK-04915: [patch] ooh323c: chan_h323 crashes on load |
ASTERISK-04916: [patch] app_sms does not receive SMS from Swisscom fixed line SMSC |
ASTERISK-04917: When chan_iax2.c is compiled with NEWJB commented out, the module will fail to link. |
ASTERISK-04918: app_rpt.so in apps/Makefile is mis-classified as obsolete, S/B experimental |
ASTERISK-04919: [patch] Fix voicemail memory leak on reload |
ASTERISK-04920: Roundrobin Queus Strategy |
ASTERISK-04921: Problems with negative value expressions. |
ASTERISK-04922: [patch] ast_mutex_lock() in ast_hangup() causes race/deadlock grief for channel drivers |
ASTERISK-04923: [patch][post 1.2] app_authenticate maxdigits option |
ASTERISK-04924: even though type=user is set in sip.conf incoming connection context finding fails |
ASTERISK-04925: [patch] [post 1.2] Simple SQL queries from the dialplan |
ASTERISK-04926: [patch] RES_ODBC Sanity Check [back from the grave] |
ASTERISK-04927: apps/Makefile cleanup |
ASTERISK-04928: Not working when dialing from ISDN BRI modem |
ASTERISK-04929: Call initiated with call file always redials |
ASTERISK-04930: A little cleanup to the main Makefile |
ASTERISK-04931: SIP notify returns gibberish |
ASTERISK-04932: Crash on GROUP_COUNT function error |
ASTERISK-04933: [patch] subscriptions on snom phones no longer work, due to an error in the last minute change of bug #3644 |
ASTERISK-04934: [patch] Fixes cdr_pgsql.so compile problem using postgresql-dev 7.4.7-6sarge1 from Debian Testing 2005-08-30 |
ASTERISK-04935: New Sounds added on 8-23-2005 (Issue 4859) break asterisk-sounds 'make install' |
ASTERISK-04936: Wrong Extension reported in CDR DB.. |
ASTERISK-04937: [patch] Make IAX2 less chatty without IAX debug turned on |
ASTERISK-04938: [patch] Call limitation fixes |
ASTERISK-04939: Random garbage returned after output reading /proc/zaptel/* |
ASTERISK-04940: [patch] Keep headers for second invite when challenged for auth |
ASTERISK-04941: [patch] struct fast_originate_helper in manager.h |
ASTERISK-04942: app_curl and codec_ilbc fails to build after recent makefile changes. |
ASTERISK-04943: Request for Change of Priority |
ASTERISK-04944: [request] [post 1.2] answeronpolarityswitch |
ASTERISK-04945: The "ringing" device state is not displayed in show queue |
ASTERISK-04946: [request] [post 1.2] too many manager connections make asterisk unusable |
ASTERISK-04947: [PATCH] CONFIG_PREEMPT_RT support for zaptel |
ASTERISK-04948: Device state handling is broken |
ASTERISK-04949: Asterisk segfaults with signal 11 from within logger.c:850 |
ASTERISK-04950: ODBC SQLRowCount() handling in res/res_config_odbc.c v1.28 |
ASTERISK-04951: [patch] Simple G.722 pass-through addition |
ASTERISK-04952: [patch] fix for: Overlooked in my bug 4830: can't detect busypattern with tones longer than 1100msecs |
ASTERISK-04953: [request] [post 1.2] Make "maxlen" of queues a function of the avalaible number of agents serving the queue |
ASTERISK-04954: [patch] Using printf in the urg_handler implicated in a deadlock |
ASTERISK-04955: [patch] notifies sent out after the expiry period of the subscription have the state 'idle' |
ASTERISK-04956: [patch] fix for potential segfault |
ASTERISK-04957: [branch] It's time to start getting T.38 into * |
ASTERISK-04958: [patch] ast_channel_inherit_variables copys variable list from parent to child in incorrect order |
ASTERISK-04959: [patch] ast_var_assign does not initialize ast_var_t data structure |
ASTERISK-04960: [patch] Send OSP token to non-OSP device |
ASTERISK-04961: [patch] avoid warning in res_odbc.c |
ASTERISK-04962: [patch] Send wrong Call-ID to OSP server |
ASTERISK-04963: [patch] [post 1.2] Dynamic calculation of queue maximum length |
ASTERISK-04964: zt_indicate warning when leaving voicemail |
ASTERISK-04965: DNS SRV lookups fail using BIND lightweight resolver |
ASTERISK-04966: [patch] Add duration limit from OSP AuthorizationResponse message |
ASTERISK-04967: [patch] service level description is inaccurate in queues.conf.sample |
ASTERISK-04968: [patch] [post 1.2] Web GUI backend for review |
ASTERISK-04969: IAX peer from database results in auto-congestion |
ASTERISK-04970: [patch] subscribe authentication fails with multiple peers on same client |
ASTERISK-04971: [patch] gcc4 warnings for iax2 |
ASTERISK-04972: [patch] gcc4 warnings for dundi |
ASTERISK-04973: [patch] gcc4 warnings for AEL |
ASTERISK-04974: Asterisk freezes with multiple "switch=>" and IAX2 on 1.0.9 stable |
ASTERISK-04975: [patch] [post 1.2] update for the mec3 echo canceller |
ASTERISK-04976: Asterisk forcing ulaw on passthrough |
ASTERISK-04977: libiax2 cannot negotiate a registration refresh of other than 60 |
ASTERISK-04978: Asterisk fails to send to correct RTP IP address when double natted |
ASTERISK-04979: coredump in chan_zap.c (1.0.2) |
ASTERISK-04980: [patch] Asterisk crashes when macro recurses infinitely |
ASTERISK-04981: Asterisk segfaults when loading chan_sip without sip.conf (or realtime database) |
ASTERISK-04982: Makefile specifies dangerous optimizations, not easy to use etc |
ASTERISK-04983: [patch] Remove outuse |
ASTERISK-04984: Code cleanup in broke cross-compilation of codec_speex.so |
ASTERISK-04985: Macro parameters not handled properly |
ASTERISK-04986: [request] Not possible to define n+101 for apps |
ASTERISK-04987: [patch] add SYSTEMSTATUS to app_system |
ASTERISK-04988: [patch] Improve handling of transfers initiated by Asterisk |
ASTERISK-04989: [patch] Change registration status types to enum |
ASTERISK-04990: [patch] [post 1.2] Implement channel locking functions |
ASTERISK-04991: [patch] dialog matching bug (pedantic mode) |
ASTERISK-04992: [patch] Implement handle_response_invite |
ASTERISK-04993: [patch] [post 1.2] allow you to set the frame milliseconds on g711,g726,gsm,ilbc and g729 from rtp.conf |
ASTERISK-04994: [patch] gcc4 warnings for ADSI |
ASTERISK-04995: [patch] gcc4 warnings for app_sms |
ASTERISK-04996: [patch] option_maxcalls option will cause hung sip channels if you are over the limit when the invite comes in. |
ASTERISK-04997: [patch] Crash on app_dial with minimally specified dial parameters |
ASTERISK-04998: [patch] documentation (doxygent and comments) enhancement for chan_agent |
ASTERISK-04999: [patch] don't initialize buffers when it's not necessary |